April 2018 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [FIX][PATCH] ASoC: topology: Fix logical inversion in set_link_hw_format()
by Xiuli Pan 16 Apr '18

16 Apr '18

From: Pan Xiuli <xiuli.pan(a)linux.intel.com> The master/slave conventions are wrt. the codec. The topology code contains a logical inversion, fix to follow ASoC usage. Signed-off-by: Pan Xiuli <xiuli.pan(a)linux.intel.com> --- sound/soc/soc-topology.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 561f74d..2931248 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -2030,11 +2030,11 @@ static void set_link_hw_format(struct snd_soc_dai_link *link, /* clock masters */ bclk_master = hw_config->bclk_master; fsync_master = hw_config->fsync_master; - if (!bclk_master && !fsync_master) + if (bclk_master && fsync_master) link->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; - else if (bclk_master && !fsync_master) - link->dai_fmt |= SND_SOC_DAIFMT_CBS_CFM; else if (!bclk_master && fsync_master) + link->dai_fmt |= SND_SOC_DAIFMT_CBS_CFM; + else if (bclk_master && !fsync_master) link->dai_fmt |= SND_SOC_DAIFMT_CBM_CFS; else link->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; -- 2.7.4

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[alsa-devel] [PATCH v3 0/2] ASoC: topology: Improve parsing hw_configs
by Kirill Marinushkin 16 Apr '18

16 Apr '18

Hello Jaroslav, Takashi, Mark, This patch series is a resend of [1] and [2], rebased on top of the latest head. It was logical to resend them together. It includes 2 patches for linux + 2 patches for alsa-lib. Please have a look. Best Regards, Kirill [1] https://patchwork.kernel.org/patch/10250485/ [2] https://patchwork.kernel.org/patch/10230611/ Kirill Marinushkin (2): ASoC: topology: Fix bclk and fsync inversion in set_link_hw_format() ASoC: topology: Add missing clock gating parameter when parsing hw_configs include/uapi/sound/asoc.h | 23 ++++++++++++++++++++--- sound/soc/soc-topology.c | 19 ++++++++++++++----- 2 files changed, 34 insertions(+), 8 deletions(-) -- 2.13.6

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[alsa-devel] [PATCH] ASoC: rt5514: Add the missing register in the readable table
by Oder Chiou 16 Apr '18

16 Apr '18

The patch adds the missing register in the readable table. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5514.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index 2d79571..c33f8da 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -89,6 +89,7 @@ static const struct reg_default rt5514_reg[] = { {RT5514_PLL3_CALIB_CTRL5, 0x40220012}, {RT5514_DELAY_BUF_CTRL1, 0x7fff006a}, {RT5514_DELAY_BUF_CTRL3, 0x00000000}, + {RT5514_ASRC_IN_CTRL1, 0x00000003}, {RT5514_DOWNFILTER0_CTRL1, 0x00020c2f}, {RT5514_DOWNFILTER0_CTRL2, 0x00020c2f}, {RT5514_DOWNFILTER0_CTRL3, 0x10000362}, @@ -181,6 +182,7 @@ static bool rt5514_readable_register(struct device *dev, unsigned int reg) case RT5514_PLL3_CALIB_CTRL5: case RT5514_DELAY_BUF_CTRL1: case RT5514_DELAY_BUF_CTRL3: + case RT5514_ASRC_IN_CTRL1: case RT5514_DOWNFILTER0_CTRL1: case RT5514_DOWNFILTER0_CTRL2: case RT5514_DOWNFILTER0_CTRL3: @@ -238,6 +240,7 @@ static bool rt5514_i2c_readable_register(struct device *dev, case RT5514_DSP_MAPPING | RT5514_PLL3_CALIB_CTRL5: case RT5514_DSP_MAPPING | RT5514_DELAY_BUF_CTRL1: case RT5514_DSP_MAPPING | RT5514_DELAY_BUF_CTRL3: + case RT5514_DSP_MAPPING | RT5514_ASRC_IN_CTRL1: case RT5514_DSP_MAPPING | RT5514_DOWNFILTER0_CTRL1: case RT5514_DSP_MAPPING | RT5514_DOWNFILTER0_CTRL2: case RT5514_DSP_MAPPING | RT5514_DOWNFILTER0_CTRL3: -- 2.7.4

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[alsa-devel] [PATCH 0/2] Minor GFP_ATOMIC removal
by Takashi Iwai 16 Apr '18

16 Apr '18

Hi, this is small fixes for the old PCI drivers that still used GFP_ATOMIC by some reason. It became a nice cleanup in the end. Takashi === Takashi Iwai (2): ALSA: cmipci: Allocate with GFP_KERNEL instead of GFP_ATOMIC ALSA: emu10k1: Reduce GFP_ATOMIC allocation include/sound/emu10k1.h | 4 ++-- sound/pci/cmipci.c | 2 +- sound/pci/emu10k1/emufx.c | 9 +-------- sound/pci/emu10k1/emupcm.c | 2 +- 4 files changed, 5 insertions(+), 12 deletions(-) -- 2.16.3

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[alsa-devel] [PATCH] ASoC: fsl_esai: Add freq check in set_dai_sysclk()
by Nicolin Chen 16 Apr '18

16 Apr '18

The freq parameter indicates the physical frequency of an actual input clock or a desired frequency of an output clock for HCKT/R. It should never be passed 0. This might cause Division-by-zero. So this patch adds a check to fix it. Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- sound/soc/fsl/fsl_esai.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index da8fd98..d79e99e 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -226,6 +226,12 @@ static int fsl_esai_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, unsigned long clk_rate; int ret; + if (freq == 0) { + dev_err(dai->dev, "%sput freq of HCK%c should not be 0Hz\n", + in ? "in" : "out", tx ? 'T' : 'R'); + return -EINVAL; + } + /* Bypass divider settings if the requirement doesn't change */ if (freq == esai_priv->hck_rate[tx] && dir == esai_priv->hck_dir[tx]) return 0; -- 2.7.4

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[alsa-devel] [PATCH 1/2] ASoC: max9860: switch to using .probe_new
by Peter Rosin 16 Apr '18

16 Apr '18

Use the new probe style for i2c drivers. Signed-off-by: Peter Rosin <peda(a)axentia.se> --- sound/soc/codecs/max9860.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/max9860.c b/sound/soc/codecs/max9860.c index a2dc6a47f466..376bd48e4790 100644 --- a/sound/soc/codecs/max9860.c +++ b/sound/soc/codecs/max9860.c @@ -599,8 +599,7 @@ static const struct dev_pm_ops max9860_pm_ops = { SET_RUNTIME_PM_OPS(max9860_suspend, max9860_resume, NULL) }; -static int max9860_probe(struct i2c_client *i2c, - const struct i2c_device_id *id) +static int max9860_probe(struct i2c_client *i2c) { struct device *dev = &i2c->dev; struct max9860_priv *max9860; @@ -738,7 +737,7 @@ static const struct of_device_id max9860_of_match[] = { MODULE_DEVICE_TABLE(of, max9860_of_match); static struct i2c_driver max9860_i2c_driver = { - .probe = max9860_probe, + .probe_new = max9860_probe, .remove = max9860_remove, .id_table = max9860_i2c_id, .driver = { -- 2.11.0

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[alsa-devel] [v3] ASoC: AMD: Support headset button on Stoney DA7219
by Akshu Agrawal 16 Apr '18

16 Apr '18

Adds headset button support. TEST=Tested Volume UP/Down, Play/Pause functionality Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- v2: Changed KEY_MEDIA to KEY_PLAYPAUSE v3: Using cz_jack.jack instead of jack->jack sound/soc/amd/acp-da7219-max98357a.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index b205c78..d281c22 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -33,6 +33,7 @@ #include <linux/gpio.h> #include <linux/module.h> #include <linux/i2c.h> +#include <linux/input.h> #include <linux/acpi.h> #include "../codecs/da7219.h" @@ -80,6 +81,11 @@ static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) return ret; } + snd_jack_set_key(cz_jack.jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + snd_jack_set_key(cz_jack.jack, SND_JACK_BTN_1, KEY_VOLUMEUP); + snd_jack_set_key(cz_jack.jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + snd_jack_set_key(cz_jack.jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); + da7219_aad_jack_det(component, &cz_jack); return 0; -- 1.9.1

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[alsa-devel] [PATCH] ASoC: rsnd: Enable IPMMU v2
by Kuninori Morimoto 16 Apr '18

16 Apr '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit 4821d914fe747 ("ASoC: rsnd: use dma_sync_single_for_xxx() for IOMMU") (= v1) which have been already reverted had supported IPMMU support on rsnd driver. Because memory allocating timing and DMAEngine access timing were different, it used continuous memory and called dma map function by itself. OTOH, it is using DMA descriptor mode (= DMA cyclic mode), thus, there was timing conflict between DMA sync/unsync and DMA transfer starting, and it maked sound noise. This patch supports IPMMU with coherent memory, and, it uses Audio DMAC dev for allocating memory by snd_pcm_lib_preallocate_pages_for_all() to indicate memory area to IPMMU. One note is that Playback/Capture need each paired Audio DMAC dev. Because of this, we need to keep each paired Audio DMAC dev when probing, and use it when allocating each memory for IPMMU. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 47 ++++++++++++++++++++++++++++++++++++++++++----- sound/soc/sh/rcar/dma.c | 7 +++++++ sound/soc/sh/rcar/rsnd.h | 1 + 3 files changed, 50 insertions(+), 5 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 4995e73..af04d41 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1352,6 +1352,37 @@ int rsnd_kctrl_new(struct rsnd_mod *mod, #define PREALLOC_BUFFER (32 * 1024) #define PREALLOC_BUFFER_MAX (32 * 1024) +static int rsnd_preallocate_pages(struct snd_soc_pcm_runtime *rtd, + struct rsnd_dai_stream *io, + int stream) +{ + struct rsnd_priv *priv = rsnd_io_to_priv(io); + struct device *dev = rsnd_priv_to_dev(priv); + struct snd_pcm_substream *substream; + int err; + + /* + * use Audio-DMAC dev if we can use IPMMU + * see + * rsnd_dmaen_attach() + */ + if (io->dmac_dev) + dev = io->dmac_dev; + + for (substream = rtd->pcm->streams[stream].substream; + substream; + substream = substream->next) { + err = snd_pcm_lib_preallocate_pages(substream, + SNDRV_DMA_TYPE_DEV, + dev, + PREALLOC_BUFFER, PREALLOC_BUFFER_MAX); + if (err < 0) + return err; + } + + return 0; +} + static int rsnd_pcm_new(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dai *dai = rtd->cpu_dai; @@ -1366,11 +1397,17 @@ static int rsnd_pcm_new(struct snd_soc_pcm_runtime *rtd) if (ret) return ret; - return snd_pcm_lib_preallocate_pages_for_all( - rtd->pcm, - SNDRV_DMA_TYPE_DEV, - rtd->card->snd_card->dev, - PREALLOC_BUFFER, PREALLOC_BUFFER_MAX); + ret = rsnd_preallocate_pages(rtd, &rdai->playback, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + return ret; + + ret = rsnd_preallocate_pages(rtd, &rdai->capture, + SNDRV_PCM_STREAM_CAPTURE); + if (ret) + return ret; + + return 0; } static const struct snd_soc_component_driver rsnd_soc_component = { diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 32ac97b..ef82b94 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -253,6 +253,13 @@ static int rsnd_dmaen_attach(struct rsnd_dai_stream *io, return -EAGAIN; } + /* + * use it for IPMMU if needed + * see + * rsnd_preallocate_pages() + */ + io->dmac_dev = chan->device->dev; + dma_release_channel(chan); dmac->dmaen_num++; diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index b1896f1..6d7280d 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -435,6 +435,7 @@ struct rsnd_dai_stream { struct snd_pcm_substream *substream; struct rsnd_mod *mod[RSND_MOD_MAX]; struct rsnd_dai *rdai; + struct device *dmac_dev; /* for IPMMU */ u32 parent_ssi_status; }; #define rsnd_io_to_mod(io, i) ((i) < RSND_MOD_MAX ? (io)->mod[(i)] : NULL) -- 1.9.1

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[alsa-devel] [PATCH] ASoC: rsnd: ssi: wait maximum 5ms for status check
by Kuninori Morimoto 16 Apr '18

16 Apr '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> It is waiting udelay(50) x 1024 (= 50ms) for status check in worst case, but it is overkill. And we shouldn't use udelay() for 50us (linux/Documentation/timers/timers-howto.txt) Waiting maximum udelay(5) x 1024 (= 5ms) is very enough for status check. This patch fixes these issue. Reported-by: Hiromitsu Yamasaki <hiromitsu.yamasaki.ym(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/ssi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 31ffe3f..9538f76 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -171,7 +171,7 @@ static void rsnd_ssi_status_check(struct rsnd_mod *mod, if (status & bit) return; - udelay(50); + udelay(5); } dev_warn(dev, "%s[%d] status check failed\n", -- 1.9.1

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[alsa-devel] [PATCH v2 1/2] ASoC: adau17x1: Handling of DSP_RUN register during fw setup
by Robert Rosengren 16 Apr '18

16 Apr '18

From: Danny Smith <dannys(a)axis.com> DSP_RUN needs to be disabled during firmware write otherwise we can end up with undefined behavior if writing to a dsp which is already running firmware. Signed-off-by: Danny Smith <dannys(a)axis.com> Signed-off-by: Robert Rosengren <robert.rosengren(a)axis.com> --- sound/soc/codecs/adau17x1.c | 26 ++++++++++++++++++++------ sound/soc/codecs/adau17x1.h | 3 ++- 2 files changed, 22 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index 80c2a06285bb..12bf24c26818 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -502,7 +502,7 @@ static int adau17x1_hw_params(struct snd_pcm_substream *substream, } if (adau->sigmadsp) { - ret = adau17x1_setup_firmware(adau, params_rate(params)); + ret = adau17x1_setup_firmware(component, params_rate(params)); if (ret < 0) return ret; } @@ -835,26 +835,40 @@ bool adau17x1_volatile_register(struct device *dev, unsigned int reg) } EXPORT_SYMBOL_GPL(adau17x1_volatile_register); -int adau17x1_setup_firmware(struct adau *adau, unsigned int rate) +int adau17x1_setup_firmware(struct snd_soc_component *component, + unsigned int rate) { int ret; - int dspsr; + int dspsr, dsp_run; + struct adau *adau = snd_soc_component_get_drvdata(component); + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + + snd_soc_dapm_mutex_lock(dapm); ret = regmap_read(adau->regmap, ADAU17X1_DSP_SAMPLING_RATE, &dspsr); if (ret) - return ret; + goto err; + + ret = regmap_read(adau->regmap, ADAU17X1_DSP_RUN, &dsp_run); + if (ret) + goto err; regmap_write(adau->regmap, ADAU17X1_DSP_ENABLE, 1); regmap_write(adau->regmap, ADAU17X1_DSP_SAMPLING_RATE, 0xf); + regmap_write(adau->regmap, ADAU17X1_DSP_RUN, 0); ret = sigmadsp_setup(adau->sigmadsp, rate); if (ret) { regmap_write(adau->regmap, ADAU17X1_DSP_ENABLE, 0); - return ret; + goto err; } regmap_write(adau->regmap, ADAU17X1_DSP_SAMPLING_RATE, dspsr); + regmap_write(adau->regmap, ADAU17X1_DSP_RUN, dsp_run); - return 0; +err: + snd_soc_dapm_mutex_unlock(dapm); + + return ret; } EXPORT_SYMBOL_GPL(adau17x1_setup_firmware); diff --git a/sound/soc/codecs/adau17x1.h b/sound/soc/codecs/adau17x1.h index a7b1cb770814..e6fe87beec07 100644 --- a/sound/soc/codecs/adau17x1.h +++ b/sound/soc/codecs/adau17x1.h @@ -68,7 +68,8 @@ int adau17x1_resume(struct snd_soc_component *component); extern const struct snd_soc_dai_ops adau17x1_dai_ops; -int adau17x1_setup_firmware(struct adau *adau, unsigned int rate); +int adau17x1_setup_firmware(struct snd_soc_component *component, + unsigned int rate); bool adau17x1_has_dsp(struct adau *adau); #define ADAU17X1_CLOCK_CONTROL 0x4000 -- 2.11.0

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